[Freeswitch-users] Freeswitch doesnt transcode

Colton Conor colton.conor at gmail.com
Mon Feb 15 19:29:40 MSK 2016


True,

But freeswitch talking to the carriers is almost always UPD.

However, freeswitch talking to the clients I would say TCP would be idea.
So its almost like freeswitch is trancoding from TCP to UDP too :)

On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice at freeswitch.org> wrote:

> The problem isn’t necessarily the devices, but there is also the carriers…
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colton Conor
> *Sent:* Monday, February 15, 2016 10:04 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch doesnt transcode
>
>
>
> So if the device supports TCP, is there any reason not to use TCP. AKA is
> there any reason to keep on using UDP. TCP seems superior.
>
>
>
> On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike at jerris.com> wrote:
>
> any device that even remotely follows sip
>
> specs supports TCP.  Most phones I have seen do.
>
>
>
> On Sunday, February 14, 2016, Colton Conor <colton.conor at gmail.com> wrote:
>
> So is TCP the preferred method of doing SIP these days? I like TCP with
> endpoints as they always break through firewalls and we never seem to have
> in issue with TCP. However UDP is a headache. So if you have the choice why
> not do TCP? I realize some devices only support UDP, but the majority of
> SIP phones out there today do support TCP.
>
>
>
> Plus if you use TLS for encryption and security then you are already using
> TCP right?
>
>
>
> On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> This behavior changed a while ago. This was dictates by ever growing SDPs
> and exceeding MTUs causing udp fragmentation. Udp does not deal with
> fragmentation and everyone refuses to fully implement sip over tcp for some
> reason even tho a ton of things support it and the RFCs require it
>
> Sent from my iPhone
>
>
> > On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
> >
> > Thanks, after setting media_mix_inbound_outbound_codecs=true,
> > transcoding happens automatically.  I remember not setting this
> > variable in other installations and transcoding used to work out of
> > the box. Is media_mix_inbound_outbound_codecs=true default in
> > Freeswitch?
> >
> >> On 14 February 2016 at 13:56, Russell Treleaven <
> rtreleaven at bunnykick.ca> wrote:
> >> fyi
> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
> >>
> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s at gmail.com>
> wrote:
> >>>
> >>> The siptrace is at http://pastebin.com/xiGqtj1Y
> >>>
> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec)
> >>> to 208 (pjsua test client with PCMU codec). The error is on line 545.
> >>>
> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com>
> >>> wrote:
> >>>> How you originate the call? Is a bridge? >From which phone?
> >>>>
> >>>> Also, please pastebin the complete sip trace (from start of leg A to
> end
> >>>> of
> >>>> both legs) and put here a link to pastebin
> >>>>
> >>>> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha
> scritto:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
> >>>>> When I call to this remote sip phone i get a 406 error that opus is
> >>>>> not supported as shown by the sip trace below. However, if I force
> the
> >>>>> codec to absolute like this
> >>>>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
> >>>>> the call works fine.
> >>>>>
> >>>>> Is there anyway I can make FreeSWITCH to automatically transcode
> >>>>> without forcing the codec string in the dial plan?
> >>>>>
> >>>>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA
> and
> >>>>> outbound_codec_prefs=PCMU,PCMA,GSM
> >>>>>
> >>>>> ---------------------------siptrace--------------------------------
> >>>>>
> >>>>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>   SIP/2.0 406 Not Acceptable
> >>>>>   Via: SIP/2.0/UDP
> >>>>>
> >>>>>
> >>>>>
> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   CSeq: 87372504 INVITE
> >>>>>   Content-Length:  0
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>>   ACK sip:303 at 192.168.1.5 SIP/2.0
> >>>>>   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
> >>>>>   Max-Forwards: 68
> >>>>>   From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
> >>>>>   To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
> >>>>>   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
> >>>>>   CSeq: 87372504 ACK
> >>>>>   Content-Length: 0
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> ------------------------------------------------------------------------
> >>>>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
> >>>>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
> >>>>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
> >>>>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
> >>>>> [SERVICE_NOT_IMPLEMENTED]
> >>>>>
> >>>>> Thanks
> >>>>> Rajil
> >>>>>
> >>>>>
> >>>>>
> _________________________________________________________________________
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> >>>>>
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> >>>>
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
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> >>>>
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> >>>
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> >>
> >>
> >>
> >>
> _________________________________________________________________________
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> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
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> >> http://www.cluecon.com
> >>
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> >
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> > http://www.freeswitchsolutions.com
> >
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> >
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
>
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>
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>
>
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>
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